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python tests accordingly.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@5065 74dad513-b988-da41-8d7b-12977e46ad98
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- Fixed no audio bug when pjsua with SRTP optional-with-duplicated-offer calls pjsua with SRTP disabled, by updating active media index after SDP negotiation done.
- Fixed bug in generating SDP, pjsua_media_channel_create_sdp(), by making sure all media in the SDP candidate are aligned with current active SDP before calling pjmedia_transport_encode_sdp().
- Fixed bug in modifying SDP for call hold, the media index to be modified was hardcoded to 0, should be active media index.
- Added python tests for calls with SRTP optional-with-duplicated-offer.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3334 74dad513-b988-da41-8d7b-12977e46ad98
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- added new PJSUA API: pjsua_verify_url() which can be used for tel: URI
- modified and tested according to spec
- added new PJSIP error code, PJSIP_ENOROUTESET, to indicate that route set is needed to send to tel: URI
- added couple of unit tests (we can't cover the whole tel: URI scenario yet)
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3323 74dad513-b988-da41-8d7b-12977e46ad98
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the placeholder for future developed tests
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2392 74dad513-b988-da41-8d7b-12977e46ad98
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